1. The Field of the Invention
The present invention generally relates to voice-over-IP telephony systems. More particularly, the present invention relates to systems and methods for detecting and analyzing voice, video, and data transmissions made via voice-over-IP systems regardless of whether signaling information relating to the transmission is present.
2. The Related Technology
Voice-over-IP (“VoIP”) telephony systems are increasingly used to enable real-time transmission of voice and video data via the Internet and associated networks. As VoIP use widens, so does the need to monitor and analyze voice, video, and data transmissions, jointly referred to here as “calls,” that are executed by VoIP systems. Indeed, a network administrator may find it useful or necessary to monitor and analyze VoIP calls taking place within the network to ensure that optimum performance is achieved, or alternatively, that problem conditions existing with respect to VoIP transactions are properly diagnosed and corrected.
Various network monitoring and analysis packages are currently available to assist a network administrator in evaluating the health of the network's VoIP system. Many of these packages combine hardware and software components to enable VoIP analysis in a number of ways, including current real-time data capture, capture buffering, and post capture analysis via saved capture files. Regardless of the capture mechanism, these monitoring and analysis packages often operate in a similar manner. VoIP calls typically include a number of voice, video, and/or other data packets, called media packets, that are sandwiched between a set of call signaling packets. The call signaling packets respectively indicate the beginning and ending of a VoIP call. Media and call signaling packets corresponding to a single call can be carried on one or more channels within the network.
A VoIP network monitor analyzes data streams within the network in search of call signaling packets indicating a VoIP call. When found, the monitor captures both the signaling packets and the interposed media packets containing the call data. A call record is then created, comprising a summary of each captured VoIP call including various metrics and basic information associated with the call, such as the call start/stop time, caller/call receiver IP addresses, call status, number of dropped (missing) packets, jitter values, and call quality. The network administrator can then review the metric data and from it determine the existence of any performance issues relating to the VoIP system operating within the network.
The media and corresponding control packets are typically transmitted using two protocols known as real-time transfer protocol (“RTP”) and real-time transfer control protocol (“RTCP”). In contrast, the call signaling packets that indicate the commencement or termination of a call can utilize a variety of independent protocols including both public standards such as H.323, SIP, and proprietary protocols such as SCCP.
Despite their utility in diagnosing VoIP system conditions, known network analyzers are sometimes unable to capture a certain number of calls. This can be due to several factors. First, occasionally the call signaling packets are transmitted within the network along a different path, or channel, than the media packets. Second, some vendors employ proprietary call signaling protocols in addition to those mentioned above that are not directly identifiable by the network analyzer. In either case, known network monitoring applications and apparatus are unable to detect a call where a call actually exists. This can result in a significant number of calls being undesirably missed by the network analyzer or related component. This can further result in an incomplete call record that in turn makes it more difficult for a network administrator to properly diagnose the network VoIP system.
In light of the above discussion, then, a need currently exists for a system by which substantially all calls in a VoIP system can be identified and captured for analysis, thereby ensuring maximum performance from VoIP systems.